mirror of
https://github.com/asterisk/asterisk.git
synced 2026-06-24 01:08:34 -07:00
Mark Michelson
10fa49e327
This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5
62 lines
2.6 KiB
Plaintext
62 lines
2.6 KiB
Plaintext
===========================================================
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===
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=== Information for upgrading between Asterisk versions
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===
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=== These files document all the changes that MUST be taken
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=== into account when upgrading between the Asterisk
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=== versions listed below. These changes may require that
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=== you modify your configuration files, dialplan or (in
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=== some cases) source code if you have your own Asterisk
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=== modules or patches. These files also include advance
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=== notice of any functionality that has been marked as
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=== 'deprecated' and may be removed in a future release,
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=== along with the suggested replacement functionality.
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===
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=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
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=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
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=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
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=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
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=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
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=== UPGRADE-11.txt -- Upgrade info for 10 to 11
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=== UPGRADE-12.txt -- Upgrade info for 11 to 12
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=== UPGRADE-13.txt -- Upgrade info for 12 to 13
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=== UPGRADE-14.txt -- Upgrade info for 13 to 14
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===========================================================
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From 14.3.0 to 14.4.0:
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res_rtp_asterisk:
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- The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
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Data and Control Packets on a Single Port." So far, the only channel driver
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that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on
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a PJSIP endpoint in pjsip.conf to enable the feature.
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New in 14.0.0
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ARI:
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- The policy for when to send "Dial" events has changed. Previously, "Dial"
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events were sent on the calling channel's topic. However, starting in Asterisk
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14, if there is no calling channel on which to send the event, the event is
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instead sent on the called channel's topic. Note that for the ARI channels
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resource's dial operation, this means that the "Dial" events will always be
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sent on the called channel's topic.
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Queue:
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- When reloading the members of a queue, the members added dynamically (i.e.
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added via the CLI command "queue add" or the AMI action "QueueAdd") now have
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their ringinuse value updated to the value of the queue. Previously, the
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ringinuse value for dynamic members was not updated on reload.
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Queue log:
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- New RINGCANCELED event is logged when the caller hangs up while ringing.
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The data1 field contains number of miliseconds since start of ringing.
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Channel Drivers:
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chan_dahdi:
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- Support for specifying a DAHDI channel using a path under /dev/dahdi
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("by name") has been removed. It was never used. Instead you should
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use kernel-level channel number allocation using span assignments.
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See the documentation of dahdi-linux and dahdi-tools.
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